On Tuesday 27 January 2009 18:06:47 pHilipp Zabel wrote: > Could you show the code? > On magician I'm just powering up uda1380 when the driver is loaded, > but I wouldn't mind saving some power :) > There are two GPIOs involved - one connected to the uda1380's RESET > line and one to control the power. > The last one doesn't have anything to do with uda1380, really. It's an > external power switch in a PMIC (or CPLD, on magican) so it would be > nice if this code could live in the machine specific drivers. > > regards > Philipp Yep, of course, here's machine driver (originally created by Denis Grigoriev) and modified codec driver. Btw, on rx1950 there're 4 gpio pins involved: GPJ0 controls codec power GPA1 controls amplifier power GPG12 is jack sense pin, it's value depends whether headphone jack is inserted or not, and GPD0 is connected to uda1380 reset.
/*
* uda1380.c - Philips UDA1380 ALSA SoC audio driver
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*
* Copyright (c) 2007 Philipp Zabel <philipp.zabel@xxxxxxxxx>
* Improved support for DAPM and audio routing/mixing capabilities,
;* added TLV support.
*
* Modified by Richard Purdie <richard@xxxxxxxxxxxxxx> to fit into SoC
* codec model.
*
* Copyright (c) 2005 Giorgio Padrin <giorgio@xxxxxxxxxxxxxxxxx>
* Copyright 2005 Openedhand Ltd.
*/
#include <linux/module.h>
#include <linux/init.h>
#include <linux/types.h>
#include <linux/string.h>
#include <linux/slab.h>
#include <linux/errno.h>
#include <linux/ioctl.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/initval.h>
#include <sound/info.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
#include "uda1380.h"
#define UDA1380_VERSION "0.6"
/*
* uda1380 register cache
*/
static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
0x0502, 0x0000, 0x0000, 0x3f3f,
0x0202, 0x0000, 0x0000, 0x0000,
0x0000, 0x0000, 0x0000, 0x0000,
0x0000, 0x0000, 0x0000, 0x0000,
0x0000, 0xff00, 0x0000, 0x4800,
0x0000, 0x0000, 0x0000, 0x0000,
0x0000, 0x0000, 0x0000, 0x0000,
0x0000, 0x0000, 0x0000, 0x0000,
0x0000, 0x8000, 0x0002, 0x0000,
};
/*
* read uda1380 register cache
*/
static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec,
unsigned int reg)
{
u16 *cache = codec->reg_cache;
if (reg == UDA1380_RESET)
return 0;
if (reg >= UDA1380_CACHEREGNUM)
return -1;
return cache[reg];
}
/*
* write uda1380 register cache
*/
static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec,
u16 reg, unsigned int value)
{
u16 *cache = codec->reg_cache;
if (reg >= UDA1380_CACHEREGNUM)
return;
cache[reg] = value;
}
/*
* write to the UDA1380 register space
*/
static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
u8 data[3];
/* data is
* data[0] is register offset
* data[1] is MS byte
* data[2] is LS byte
*/
data[0] = reg;
data[1] = (value & 0xff00) >> 8;
data[2] = value & 0x00ff;
uda1380_write_reg_cache(codec, reg, value);
/* the interpolator & decimator regs must only be written when the
* codec DAI is active.
*/
if (!codec->active && (reg >= UDA1380_MVOL))
return 0;
pr_debug("uda1380: hw write %x val %x\n", reg, value);
if (codec->hw_write(codec->control_data, data, 3) == 3) {
unsigned int val;
i2c_master_send(codec->control_data, data, 1);
i2c_master_recv(codec->control_data, data, 2);
val = (data[0]<<8) | data[1];
if (val != value) {
pr_debug("uda1380: READ BACK VAL %x\n",
(data[0]<<8) | data[1]);
return -EIO;
}
return 0;
} else
return -EIO;
}
#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0)
/* declarations of ALSA reg_elem_REAL controls */
static const char *uda1380_deemp[] = {
"None",
"32kHz",
"44.1kHz",
"48kHz",
"96kHz",
};
static const char *uda1380_input_sel[] = {
"Line",
"Mic + Line R",
"Line L",
"Mic",
};
static const char *uda1380_output_sel[] = {
"DAC",
"Analog Mixer",
};
static const char *uda1380_spf_mode[] = {
"Flat",
"Minimum1",
"Minimum2",
"Maximum"
};
static const char *uda1380_capture_sel[] = {
"ADC",
"Digital Mixer"
};
static const char *uda1380_sel_ns[] = {
"3rd-order",
"5th-order"
};
static const char *uda1380_mix_control[] = {
"off",
"PCM only",
"before sound processing",
"after sound processing"
};
static const char *uda1380_sdet_setting[] = {
"3200",
"4800",
"9600",
"19200"
};
static const char *uda1380_os_setting[] = {
"single-speed",
"double-speed (no mixing)",
"quad-speed (no mixing)"
};
static const struct soc_enum uda1380_deemp_enum[] = {
SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp),
SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp),
};
static const struct soc_enum uda1380_input_sel_enum =
SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */
static const struct soc_enum uda1380_output_sel_enum =
SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */
static const struct soc_enum uda1380_spf_enum =
SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */
static const struct soc_enum uda1380_capture_sel_enum =
SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */
static const struct soc_enum uda1380_sel_ns_enum =
SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */
static const struct soc_enum uda1380_mix_enum =
SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */
static const struct soc_enum uda1380_sdet_enum =
SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */
static const struct soc_enum uda1380_os_enum =
SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */
/*
* from -48 dB in 1.5 dB steps (mute instead of -49.5 dB)
*/
static DECLARE_TLV_DB_SCALE(amix_tlv, -4950, 150, 1);
/*
* from -78 dB in 1 dB steps (3 dB steps, really. LSB are ignored),
* from -66 dB in 0.5 dB steps (2 dB steps, really) and
* from -52 dB in 0.25 dB steps
*/
static const unsigned int mvol_tlv[] = {
TLV_DB_RANGE_HEAD(3),
0, 15, TLV_DB_SCALE_ITEM(-8200, 100, 1),
16, 43, TLV_DB_SCALE_ITEM(-6600, 50, 0),
44, 252, TLV_DB_SCALE_ITEM(-5200, 25, 0),
};
/*
* from -72 dB in 1.5 dB steps (6 dB steps really),
* from -66 dB in 0.75 dB steps (3 dB steps really),
* from -60 dB in 0.5 dB steps (2 dB steps really) and
* from -46 dB in 0.25 dB steps
*/
static const unsigned int vc_tlv[] = {
TLV_DB_RANGE_HEAD(4),
0, 7, TLV_DB_SCALE_ITEM(-7800, 150, 1),
8, 15, TLV_DB_SCALE_ITEM(-6600, 75, 0),
16, 43, TLV_DB_SCALE_ITEM(-6000, 50, 0),
44, 228, TLV_DB_SCALE_ITEM(-4600, 25, 0),
};
/* from 0 to 6 dB in 2 dB steps if SPF mode != flat */
static DECLARE_TLV_DB_SCALE(tr_tlv, 0, 200, 0);
/* from 0 to 24 dB in 2 dB steps, if SPF mode == maximum, otherwise cuts
* off at 18 dB max) */
static DECLARE_TLV_DB_SCALE(bb_tlv, 0, 200, 0);
/* from -63 to 24 dB in 0.5 dB steps (-128...48) */
static DECLARE_TLV_DB_SCALE(dec_tlv, -6400, 50, 1);
/* from 0 to 24 dB in 3 dB steps */
static DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0);
/* from 0 to 30 dB in 2 dB steps */
static DECLARE_TLV_DB_SCALE(vga_tlv, 0, 200, 0);
static const struct snd_kcontrol_new uda1380_snd_controls[] = {
SOC_DOUBLE_TLV("Analog Mixer Volume", UDA1380_AMIX, 0, 8, 44, 1, amix_tlv), /* AVCR, AVCL */
SOC_DOUBLE_TLV("Master Playback Volume", UDA1380_MVOL, 0, 8, 252, 1, mvol_tlv), /* MVCL, MVCR */
SOC_SINGLE_TLV("ADC Playback Volume", UDA1380_MIXVOL, 8, 228, 1, vc_tlv), /* VC2 */
SOC_SINGLE_TLV("PCM Playback Volume", UDA1380_MIXVOL, 0, 228, 1, vc_tlv), /* VC1 */
SOC_ENUM("Sound Processing Filter", uda1380_spf_enum), /* M */
SOC_DOUBLE_TLV("Tone Control - Treble", UDA1380_MODE, 4, 12, 3, 0, tr_tlv), /* TRL, TRR */
SOC_DOUBLE_TLV("Tone Control - Bass", UDA1380_MODE, 0, 8, 15, 0, bb_tlv), /* BBL, BBR */
/**/ SOC_SINGLE("Master Playback Switch", UDA1380_DEEMP, 14, 1, 1), /* MTM */
SOC_SINGLE("ADC Playback Switch", UDA1380_DEEMP, 11, 1, 1), /* MT2 from decimation filter */
SOC_ENUM("ADC Playback De-emphasis", uda1380_deemp_enum[0]), /* DE2 */
SOC_SINGLE("PCM Playback Switch", UDA1380_DEEMP, 3, 1, 1), /* MT1, from digital data input */
SOC_ENUM("PCM Playback De-emphasis", uda1380_deemp_enum[1]), /* DE1 */
SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */
SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */
SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */
SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */
SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */
SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */
SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */
SOC_DOUBLE_S8_TLV("ADC Capture Volume", UDA1380_DEC, -128, 48, dec_tlv), /* ML_DEC, MR_DEC */
/**/ SOC_SINGLE("ADC Capture Switch", UDA1380_PGA, 15, 1, 1), /* MT_ADC */
SOC_DOUBLE_TLV("Line Capture Volume", UDA1380_PGA, 0, 8, 8, 0, pga_tlv), /* PGA_GAINCTRLL, PGA_GAINCTRLR */
SOC_SINGLE("ADC Polarity inverting Switch", UDA1380_ADC, 12, 1, 0), /* ADCPOL_INV */
SOC_SINGLE_TLV("Mic Capture Volume", UDA1380_ADC, 8, 15, 0, vga_tlv), /* VGA_CTRL */
SOC_SINGLE("DC Filter Bypass Switch", UDA1380_ADC, 1, 1, 0), /* SKIP_DCFIL (before decimator) */
SOC_SINGLE("DC Filter Enable Switch", UDA1380_ADC, 0, 1, 0), /* EN_DCFIL (at output of decimator) */
SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), /* TODO: enum, see table 62 */
SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), /* AGC_LEVEL */
/* -5.5, -8, -11.5, -14 dBFS */
SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0),
};
/* add non dapm controls */
static int uda1380_add_controls(struct snd_soc_codec *codec)
{
int err, i;
for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) {
err = snd_ctl_add(codec->card,
snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL));
if (err < 0)
return err;
}
return 0;
}
/* Input mux */
static const struct snd_kcontrol_new uda1380_input_mux_control =
SOC_DAPM_ENUM("Route", uda1380_input_sel_enum);
/* Output mux */
static const struct snd_kcontrol_new uda1380_output_mux_control =
SOC_DAPM_ENUM("Route", uda1380_output_sel_enum);
/* Capture mux */
static const struct snd_kcontrol_new uda1380_capture_mux_control =
SOC_DAPM_ENUM("Route", uda1380_capture_sel_enum);
static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
&uda1380_input_mux_control),
SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0,
&uda1380_output_mux_control),
SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0,
&uda1380_capture_mux_control),
SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0),
SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0),
SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0),
SND_SOC_DAPM_INPUT("VINM"),
SND_SOC_DAPM_INPUT("VINL"),
SND_SOC_DAPM_INPUT("VINR"),
SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("VOUTLHP"),
SND_SOC_DAPM_OUTPUT("VOUTRHP"),
SND_SOC_DAPM_OUTPUT("VOUTL"),
SND_SOC_DAPM_OUTPUT("VOUTR"),
SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0),
SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0),
};
static const struct snd_soc_dapm_route audio_map[] = {
/* output mux */
{"HeadPhone Driver", NULL, "Output Mux"},
{"VOUTR", NULL, "Output Mux"},
{"VOUTL", NULL, "Output Mux"},
{"Analog Mixer", NULL, "VINR"},
{"Analog Mixer", NULL, "VINL"},
{"Analog Mixer", NULL, "DAC"},
{"Output Mux", "DAC", "DAC"},
{"Output Mux", "Analog Mixer", "Analog Mixer"},
/* {"DAC", "Digital Mixer", "I2S" } */
/* headphone driver */
{"VOUTLHP", NULL, "HeadPhone Driver"},
{"VOUTRHP", NULL, "HeadPhone Driver"},
/* input mux */
{"Left ADC", NULL, "Input Mux"},
{"Input Mux", "Mic", "Mic LNA"},
{"Input Mux", "Mic + Line R", "Mic LNA"},
{"Input Mux", "Line L", "Left PGA"},
{"Input Mux", "Line", "Left PGA"},
/* right input */
{"Right ADC", "Mic + Line R", "Right PGA"},
{"Right ADC", "Line", "Right PGA"},
/* inputs */
{"Mic LNA", NULL, "VINM"},
{"Left PGA", NULL, "VINL"},
{"Right PGA", NULL, "VINR"},
};
static int uda1380_add_widgets(struct snd_soc_codec *codec)
{
snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
ARRAY_SIZE(uda1380_dapm_widgets));
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
snd_soc_dapm_new_widgets(codec);
return 0;
}
static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
int iface;
/* set up DAI based upon fmt */
iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK);
/* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
iface |= R01_SFORI_I2S | R01_SFORO_I2S;
break;
case SND_SOC_DAIFMT_LSB:
iface |= R01_SFORI_LSB16 | R01_SFORO_I2S;
break;
case SND_SOC_DAIFMT_MSB:
iface |= R01_SFORI_MSB | R01_SFORO_I2S;
}
if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
iface |= R01_SIM;
uda1380_write(codec, UDA1380_IFACE, iface);
return 0;
}
/*
* Flush reg cache
* We can only write the interpolator and decimator registers
* when the DAI is being clocked by the CPU DAI. It's up to the
* machine and cpu DAI driver to do this before we are called.
*/
static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
int reg, reg_start, reg_end, clk;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
reg_start = UDA1380_MVOL;
reg_end = UDA1380_MIXER;
} else {
reg_start = UDA1380_DEC;
reg_end = UDA1380_AGC;
}
/* FIXME disable DAC_CLK */
clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);
for (reg = reg_start; reg <= reg_end; reg++) {
pr_debug("uda1380: flush reg %x val %x:", reg,
uda1380_read_reg_cache(codec, reg));
uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg));
}
/* FIXME enable DAC_CLK */
uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK);
return 0;
}
static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
/* set WSPLL power and divider if running from this clock */
if (clk & R00_DAC_CLK) {
int rate = params_rate(params);
u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
clk &= ~0x3; /* clear SEL_LOOP_DIV */
switch (rate) {
case 6250 ... 12500:
clk |= 0x0;
break;
case 12501 ... 25000:
clk |= 0x1;
break;
case 25001 ... 50000:
clk |= 0x2;
break;
case 50001 ... 100000:
clk |= 0x3;
break;
}
uda1380_write(codec, UDA1380_PM, R02_PON_PLL | pm);
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
clk |= R00_EN_DAC | R00_EN_INT;
else
clk |= R00_EN_ADC | R00_EN_DEC;
uda1380_write(codec, UDA1380_CLK, clk);
return 0;
}
static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
/* shut down WSPLL power if running from this clock */
if (clk & R00_DAC_CLK) {
u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
uda1380_write(codec, UDA1380_PM, ~R02_PON_PLL & pm);
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
clk &= ~(R00_EN_DAC | R00_EN_INT);
else
clk &= ~(R00_EN_ADC | R00_EN_DEC);
uda1380_write(codec, UDA1380_CLK, clk);
}
static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM;
/* FIXME: mute(codec,0) is called when the magician clock is already
* set to WSPLL, but for some unknown reason writing to interpolator
* registers works only when clocked by SYSCLK */
u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
if (mute)
uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM);
else
uda1380_write(codec, UDA1380_DEEMP, mute_reg);
uda1380_write(codec, UDA1380_CLK, clk);
return 0;
}
static int uda1380_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
int pm = uda1380_read_reg_cache(codec, UDA1380_PM);
void (*power)(int) = codec->private_data;
int i;
u8 *cache = codec->reg_cache;
switch (level) {
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
power(1);
uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm);
break;
case SND_SOC_BIAS_STANDBY:
if (power) power(1);
uda1380_write(codec, UDA1380_PM, R02_PON_BIAS);
break;
case SND_SOC_BIAS_OFF:
uda1380_write(codec, UDA1380_PM, 0x0);
if (power) power(0);
break;
}
codec->bias_level = level;
return 0;
}
#define UDA1380_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
struct snd_soc_dai uda1380_dai[] = {
{
.name = "UDA1380",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = UDA1380_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = UDA1380_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.ops = {
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
.prepare = uda1380_pcm_prepare,
},
.dai_ops = {
.digital_mute = uda1380_mute,
.set_fmt = uda1380_set_dai_fmt,
},
},
{ /* playback only - dual interface */
.name = "UDA1380",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = UDA1380_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.ops = {
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
.prepare = uda1380_pcm_prepare,
},
.dai_ops = {
.digital_mute = uda1380_mute,
.set_fmt = uda1380_set_dai_fmt,
},
},
{ /* capture only - dual interface*/
.name = "UDA1380",
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = UDA1380_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.ops = {
.hw_params = uda1380_pcm_hw_params,
.shutdown = uda1380_pcm_shutdown,
.prepare = uda1380_pcm_prepare,
},
.dai_ops = {
.set_fmt = uda1380_set_dai_fmt,
},
},
};
EXPORT_SYMBOL_GPL(uda1380_dai);
static int uda1380_suspend(struct platform_device *pdev, pm_message_t state)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int uda1380_resume(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
int i;
u8 data[2];
u16 *cache = codec->reg_cache;
/* Sync reg_cache with the hardware */
for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) {
data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
data[1] = cache[i] & 0x00ff;
codec->hw_write(codec->control_data, data, 2);
}
uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
uda1380_set_bias_level(codec, codec->suspend_bias_level);
return 0;
}
/*
* initialise the UDA1380 driver
* register mixer and dsp interfaces with the kernel
*/
static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
{
struct snd_soc_codec *codec = socdev->codec;
int ret = 0;
codec->name = "UDA1380";
codec->owner = THIS_MODULE;
codec->read = uda1380_read_reg_cache;
codec->write = uda1380_write;
codec->set_bias_level = uda1380_set_bias_level;
codec->dai = uda1380_dai;
codec->num_dai = ARRAY_SIZE(uda1380_dai);
codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg),
GFP_KERNEL);
if (codec->reg_cache == NULL)
return -ENOMEM;
codec->reg_cache_size = ARRAY_SIZE(uda1380_reg);
codec->reg_cache_step = 1;
uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
uda1380_reset(codec);
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
pr_err("uda1380: failed to create pcms\n");
goto pcm_err;
}
/* power on device */
/* set clock input */
switch (dac_clk) {
case UDA1380_DAC_CLK_SYSCLK:
uda1380_write(codec, UDA1380_CLK, 0);
break;
case UDA1380_DAC_CLK_WSPLL:
uda1380_write(codec, UDA1380_CLK, R00_DAC_CLK);
break;
}
/* uda1380 init */
uda1380_add_controls(codec);
uda1380_add_widgets(codec);
ret = snd_soc_register_card(socdev);
if (ret < 0) {
pr_err("uda1380: failed to register card\n");
goto card_err;
}
return ret;
card_err:
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
pcm_err:
kfree(codec->reg_cache);
return ret;
}
static struct snd_soc_device *uda1380_socdev;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static int uda1380_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct snd_soc_device *socdev = uda1380_socdev;
struct uda1380_setup_data *setup = socdev->codec_data;
struct snd_soc_codec *codec = socdev->codec;
int ret;
i2c_set_clientdata(i2c, codec);
codec->control_data = i2c;
ret = uda1380_init(socdev, setup->dac_clk);
if (ret < 0)
pr_err("uda1380: failed to initialise UDA1380\n");
return ret;
}
static int uda1380_i2c_remove(struct i2c_client *client)
{
struct snd_soc_codec *codec = i2c_get_clientdata(client);
kfree(codec->reg_cache);
return 0;
}
static const struct i2c_device_id uda1380_i2c_id[] = {
{ "uda1380", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id);
static struct i2c_driver uda1380_i2c_driver = {
.driver = {
.name = "UDA1380 I2C Codec",
.owner = THIS_MODULE,
},
.probe = uda1380_i2c_probe,
.remove = uda1380_i2c_remove,
.id_table = uda1380_i2c_id,
};
static int uda1380_add_i2c_device(struct platform_device *pdev,
const struct uda1380_setup_data *setup)
{
struct i2c_board_info info;
struct i2c_adapter *adapter;
struct i2c_client *client;
int ret;
ret = i2c_add_driver(&uda1380_i2c_driver);
if (ret != 0) {
dev_err(&pdev->dev, "can't add i2c driver\n");
return ret;
}
memset(&info, 0, sizeof(struct i2c_board_info));
info.addr = setup->i2c_address;
strlcpy(info.type, "uda1380", I2C_NAME_SIZE);
adapter = i2c_get_adapter(setup->i2c_bus);
if (!adapter) {
dev_err(&pdev->dev, "can't get i2c adapter %d\n",
setup->i2c_bus);
goto err_driver;
}
client = i2c_new_device(adapter, &info);
i2c_put_adapter(adapter);
if (!client) {
dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
(unsigned int)info.addr);
goto err_driver;
}
return 0;
err_driver:
i2c_del_driver(&uda1380_i2c_driver);
return -ENODEV;
}
#endif
static int uda1380_probe(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct uda1380_setup_data *setup;
struct snd_soc_codec *codec;
int ret;
pr_info("UDA1380 Audio Codec %s\n", UDA1380_VERSION);
setup = socdev->codec_data;
codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
if (codec == NULL)
return -ENOMEM;
socdev->codec = codec;
/* Ugly, but should work */
codec->private_data = setup->power;
mutex_init(&codec->mutex);
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
uda1380_socdev = socdev;
ret = -ENODEV;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
if (setup->i2c_address) {
codec->hw_write = (hw_write_t)i2c_master_send;
ret = uda1380_add_i2c_device(pdev, setup);
}
#endif
if (ret != 0)
kfree(codec);
return ret;
}
/* power down chip */
static int uda1380_remove(struct platform_device *pdev)
{
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
if (codec->control_data)
uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_unregister_device(codec->control_data);
i2c_del_driver(&uda1380_i2c_driver);
#endif
kfree(codec);
return 0;
}
struct snd_soc_codec_device soc_codec_dev_uda1380 = {
.probe = uda1380_probe,
.remove = uda1380_remove,
.suspend = uda1380_suspend,
.resume = uda1380_resume,
};
EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
MODULE_AUTHOR("Giorgio Padrin");
MODULE_DESCRIPTION("Audio support for codec Philips UDA1380");
MODULE_LICENSE("GPL");
/*
* rx1950.c -- ALSA Soc Audio Layer
*
* Copyright (c) 2007 Roman Moravcik <roman.moravcik@xxxxxxxxx>
*
* Based on smdk2440.c and magician.c
*
* Authors: Graeme Gregory graeme.gregory@xxxxxxxxxxxxxxxx
* Philipp Zabel <philipp.zabel@xxxxxxxxx>
* Denis Grigoriev <dgreenday@xxxxxxxxx>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/timer.h>
#include <linux/delay.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <linux/spinlock.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
#include <asm/hardware/scoop.h>
#include <asm/plat-s3c24xx/regs-iis.h>
#include <mach/regs-clock.h>
#include <mach/regs-gpio.h>
#include <mach/regs-gpioj.h>
#include <mach/audio.h>
#include <asm/io.h>
#include <mach/hardware.h>
#include "../codecs/uda1380.h"
#include "s3c24xx-pcm.h"
#include "s3c24xx-i2s.h"
//#define RX1950_DEBUG
#ifdef RX1950_DEBUG
#define DBG(x...) printk(KERN_INFO x)
#else
#define DBG(x...)
#endif
#define RX1950_HP_OFF 0
#define RX1950_HP_ON 1
#define RX1950_MIC 2
#define RX1950_SPK_ON 0
#define RX1950_SPK_OFF 1
static int rx1950_jack_func = RX1950_HP_ON;
static int rx1950_spk_func = RX1950_SPK_OFF;
static void rx1950_ext_control(struct snd_soc_codec *codec)
{
printk("%s entered\n", __func__);
if (rx1950_spk_func == RX1950_SPK_ON)
snd_soc_dapm_enable_pin(codec, "Speaker");
else
snd_soc_dapm_disable_pin(codec, "Speaker");
/* set up jack connection */
switch (rx1950_jack_func) {
case RX1950_HP_OFF:
snd_soc_dapm_disable_pin(codec, "Headphone Jack");
snd_soc_dapm_disable_pin(codec, "Mic Jack");
break;
case RX1950_HP_ON:
snd_soc_dapm_enable_pin(codec, "Headphone Jack");
snd_soc_dapm_disable_pin(codec, "Mic Jack");
break;
case RX1950_MIC:
snd_soc_dapm_disable_pin(codec, "Headphone Jack");
snd_soc_dapm_enable_pin(codec, "Mic Jack");
break;
}
snd_soc_dapm_sync(codec);
}
static int rx1950_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->socdev->codec;
printk("%s entered\n", __func__);
/* check the jack status at stream startup */
rx1950_ext_control(codec);
return 0;
}
static void rx1950_shutdown(struct snd_pcm_substream *substream)
{
// struct snd_soc_pcm_runtime *rtd = substream->private_data;
// struct snd_soc_codec *codec = rtd->socdev->codec;
//DBG("Entered rx1950_shutdown\n");
printk("%s entered\n", __func__);
}
static int rx1950_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
unsigned long iis_clkrate;
int div, div256, div384, diff256, diff384, bclk;
int ret;
unsigned int rate=params_rate(params);
DBG("Entered %s\n",__FUNCTION__);
iis_clkrate = s3c24xx_i2s_get_clockrate();
DBG("iis_clkrate = %li\n", iis_clkrate);
/* Using PCLK doesnt seem to suit audio particularly well on these cpu's
*/
div256 = iis_clkrate / (rate * 256);
div384 = iis_clkrate / (rate * 384);
if (((iis_clkrate / div256) - (rate * 256)) <
((rate * 256) - (iis_clkrate / (div256 + 1)))) {
diff256 = (iis_clkrate / div256) - (rate * 256);
} else {
div256++;
diff256 = (iis_clkrate / div256) - (rate * 256);
}
if (((iis_clkrate / div384) - (rate * 384)) <
((rate * 384) - (iis_clkrate / (div384 + 1)))) {
diff384 = (iis_clkrate / div384) - (rate * 384);
} else {
div384++;
diff384 = (iis_clkrate / div384) - (rate * 384);
}
DBG("diff256 %d, diff384 %d\n", diff256, diff384);
if (diff256<=diff384) {
DBG("Selected 256FS\n");
div = div256;
bclk = S3C2410_IISMOD_256FS;
} else {
DBG("Selected 384FS\n");
div = div384;
bclk = S3C2410_IISMOD_384FS;
}
/* set codec DAI configuration */
ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set the audio system clock for DAC and ADC */
ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK,
rate, SND_SOC_CLOCK_OUT);
if (ret < 0)
return ret;
/* set MCLK division for sample rate */
ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, S3C2410_IISMOD_32FS );
if (ret < 0)
return ret;
/* set BCLK division for sample rate */
ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, bclk);
if (ret < 0)
return ret;
/* set prescaler division for sample rate */
ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
S3C24XX_PRESCALE(div,div));
if (ret < 0)
return ret;
return 0;
}
static struct snd_soc_ops rx1950_ops = {
.startup = rx1950_startup,
.shutdown = rx1950_shutdown,
.hw_params = rx1950_hw_params,
// .prepare = rx1950_playback_prepare,
// .hw_free = rx1950_hw_free,
};
static int rx1950_get_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = rx1950_jack_func;
return 0;
}
static int rx1950_set_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
if (rx1950_jack_func == ucontrol->value.integer.value[0])
return 0;
rx1950_jack_func = ucontrol->value.integer.value[0];
rx1950_ext_control(codec);
return 1;
}
static int rx1950_get_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = rx1950_spk_func;
return 0;
}
static int rx1950_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
if (rx1950_spk_func == ucontrol->value.integer.value[0])
return 0;
rx1950_spk_func = ucontrol->value.integer.value[0];
if (rx1950_spk_func) {
s3c2410_gpio_setpin(S3C2410_GPA1, 0);
} else {
s3c2410_gpio_setpin(S3C2410_GPA1, 1);
}
rx1950_ext_control(codec);
return 1;
}
static int rx1950_spk_power(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
s3c2410_gpio_cfgpin(S3C2410_GPA1, S3C2410_GPIO_OUTPUT);
if (SND_SOC_DAPM_EVENT_ON(event) && (rx1950_spk_func == 0))
s3c2410_gpio_setpin(S3C2410_GPA1, 1);
else
s3c2410_gpio_setpin(S3C2410_GPA1, 0);
return 0;
}
/* rx1950 machine dapm widgets */
static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Mic Jack", NULL),
SND_SOC_DAPM_SPK("Speaker", rx1950_spk_power),
};
/* rx1950 machine audio_map */
static const struct snd_soc_dapm_route audio_map[] = {
/* headphone connected to VOUTLHP, VOUTRHP */
{"Headphone Jack", NULL, "VOUTLHP"},
{"Headphone Jack", NULL, "VOUTRHP"},
/* ext speaker connected to VOUTL, VOUTR */
{"Speaker", NULL, "VOUTL"},
{"Speaker", NULL, "VOUTR"},
/* mic is connected to VINM */
{"VINM", NULL, "Mic Jack"},
};
static const char *jack_function[] = {"Off", "Headphone", "Mic"};
static const char *spk_function[] = {"On", "Off"};
static const struct soc_enum rx1950_enum[] = {
SOC_ENUM_SINGLE_EXT(3, jack_function),
SOC_ENUM_SINGLE_EXT(2, spk_function),
};
static const struct snd_kcontrol_new uda1380_rx1950_controls[] = {
SOC_ENUM_EXT("Jack Function", rx1950_enum[0], rx1950_get_jack,
rx1950_set_jack),
SOC_ENUM_EXT("Speaker Function", rx1950_enum[1], rx1950_get_spk,
rx1950_set_spk),
};
/*
* Logic for a UDA1380 as attached to RX1950
*/
static int rx1950_uda1380_init(struct snd_soc_codec *codec)
{
int i, err;
DBG("Staring rx1950 init\n");
/* NC codec pins */
snd_soc_dapm_nc_pin(codec, "VOUTLHP");
snd_soc_dapm_nc_pin(codec, "VOUTRHP");
/* Add rx1950 specific widgets */
snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
ARRAY_SIZE(uda1380_dapm_widgets));
rx1950_ext_control(codec);
/* Add rx1950 specific controls */
for (i = 0; i < ARRAY_SIZE(uda1380_rx1950_controls); i++) {
if ((err = snd_ctl_add(codec->card,
snd_soc_cnew(&uda1380_rx1950_controls[i],
codec, NULL))) < 0)
return err;
}
/* Set up rx1950 specific audio path audio_mapnects */
err = snd_soc_dapm_add_routes(codec, audio_map,
ARRAY_SIZE(audio_map));
snd_soc_dapm_sync(codec);
DBG("Ending rx1950 init\n");
return 0;
}
/* s3c24xx digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link s3c24xx_dai = {
.name = "uda1380",
.stream_name = "UDA1380",
.cpu_dai = &s3c24xx_i2s_dai,
.codec_dai = &uda1380_dai[UDA1380_DAI_DUPLEX],
.init = rx1950_uda1380_init,
.ops = &rx1950_ops,
};
/* rx1950 audio machine driver */
static struct snd_soc_machine snd_soc_machine_rx1950 = {
.name = "RX1950",
.dai_link = &s3c24xx_dai,
.num_links = 1,
};
static void rx1950_codec_enable(int enable);
static struct uda1380_setup_data rx1950_uda1380_setup = {
.i2c_address = 0x1a,
.dac_clk = UDA1380_DAC_CLK_SYSCLK,
.power = rx1950_codec_enable,
};
/* s3c24xx audio subsystem */
static struct snd_soc_device s3c24xx_snd_devdata = {
.machine = &snd_soc_machine_rx1950,
.platform = &s3c24xx_soc_platform,
.codec_dev = &soc_codec_dev_uda1380,
.codec_data = &rx1950_uda1380_setup,
};
static struct platform_device *s3c24xx_snd_device;
//static DECLARE_COMPLETION(rx1950_sound_completion);
static DECLARE_MUTEX(rx1950_power_mutex);
//static spinlock_t cmpl_lock = SPIN_LOCK_UNLOCKED;
//static int waiting_for_completion;
static irqreturn_t codec_enabled(int irq, void *dev_id)
{
/*spin_lock(&cmpl_lock);
if (waiting_for_completion) {
printk("%s: completed!\n", __func__);
complete(&rx1950_sound_completion);
waiting_for_completion = 0;
}
spin_unlock(&cmpl_lock);*/
//printk("%s: got here!\n", __func__);
return IRQ_HANDLED;
}
static void rx1950_codec_enable(int enable)
{
down(&rx1950_power_mutex);
printk("%s: enable == %d\n", __func__, enable);
if (enable) {
if (s3c2410_gpio_getpin(S3C2440_GPJ0))
goto done;
//spin_lock(&cmpl_lock);
//waiting_for_completion = 1;
//spin_unlock(&cmpl_lock);
s3c2410_gpio_setpin(S3C2410_GPD0, 0);
s3c2410_gpio_setpin(S3C2440_GPJ0, 0);
s3c2410_gpio_setpin(S3C2440_GPJ0, 1);
/* Wait for EINT20 irq to ensure uda1380 is powered */
//printk("%s: waiting for compeltion...\n", __func__);
//wait_for_completion(&rx1950_sound_completion);
//printk("%s: completed\n", __func__);
//printk("%s: GPG12: %d\n", __func__,
// s3c2410_gpio_getpin(S3C2410_GPG12));
mdelay(50);
s3c2410_gpio_setpin(S3C2410_GPD0, 1);
s3c2410_gpio_setpin(S3C2410_GPD0, 0);
}
else {
if (!s3c2410_gpio_getpin(S3C2440_GPJ0))
goto done;
s3c2410_gpio_setpin(S3C2440_GPJ0, 1);
s3c2410_gpio_setpin(S3C2440_GPJ0, 0);
}
done:
printk("%s: done!\n", __func__);
up(&rx1950_power_mutex);
}
static int __init rx1950_init(void)
{
int ret;
/* configure some gpios */
s3c2410_gpio_cfgpin(S3C2410_GPD0, S3C2410_GPIO_OUTPUT);
s3c2410_gpio_cfgpin(S3C2410_GPG12, S3C2410_GPIO_IRQ);
s3c2410_gpio_cfgpin(S3C2440_GPJ0, S3C2410_GPIO_OUTPUT);
s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
if (!s3c24xx_snd_device) {
printk(KERN_ERR "platform_dev_alloc failed\n");
return -ENOMEM;
}
if (request_irq(IRQ_EINT20, codec_enabled,
IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
"snd_soc_rx1950_uda1380", s3c24xx_snd_device)) {
printk(KERN_ERR "can't get codec enabled irq.\n");
/* FIXME: fix memory leak here! */
return -ENOENT;
}
/* enable codec's power */
platform_set_drvdata(s3c24xx_snd_device, &s3c24xx_snd_devdata);
s3c24xx_snd_devdata.dev = &s3c24xx_snd_device->dev;
ret = platform_device_add(s3c24xx_snd_device);
if (ret) {
DBG("ret = %i\n",ret);
platform_device_put(s3c24xx_snd_device);
}
printk("%s done\n", __func__);
return ret;
}
static void __exit rx1950_exit(void)
{
disable_irq(IRQ_EINT20);
free_irq(IRQ_EINT20, s3c24xx_snd_device);
platform_device_unregister(s3c24xx_snd_device);
}
module_init(rx1950_init);
module_exit(rx1950_exit);
/* Module information */
MODULE_AUTHOR("Denis Grigoriev");
MODULE_DESCRIPTION("ALSA SoC RX1950");
MODULE_LICENSE("GPL");
/*
* Audio support for Philips UDA1380
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*
* Copyright (c) 2005 Giorgio Padrin <giorgio@xxxxxxxxxxxxxxxxx>
*/
#ifndef _UDA1380_H
#define _UDA1380_H
#define UDA1380_CLK 0x00
#define UDA1380_IFACE 0x01
#define UDA1380_PM 0x02
#define UDA1380_AMIX 0x03
#define UDA1380_HP 0x04
#define UDA1380_MVOL 0x10
#define UDA1380_MIXVOL 0x11
#define UDA1380_MODE 0x12
#define UDA1380_DEEMP 0x13
#define UDA1380_MIXER 0x14
#define UDA1380_INTSTAT 0x18
#define UDA1380_DEC 0x20
#define UDA1380_PGA 0x21
#define UDA1380_ADC 0x22
#define UDA1380_AGC 0x23
#define UDA1380_DECSTAT 0x28
#define UDA1380_RESET 0x7f
#define UDA1380_CACHEREGNUM 0x24
/* Register flags */
#define R00_EN_ADC 0x0800
#define R00_EN_DEC 0x0400
#define R00_EN_DAC 0x0200
#define R00_EN_INT 0x0100
#define R00_DAC_CLK 0x0010
#define R01_SFORI_I2S 0x0000
#define R01_SFORI_LSB16 0x0100
#define R01_SFORI_LSB18 0x0200
#define R01_SFORI_LSB20 0x0300
#define R01_SFORI_MSB 0x0500
#define R01_SFORI_MASK 0x0700
#define R01_SFORO_I2S 0x0000
#define R01_SFORO_LSB16 0x0001
#define R01_SFORO_LSB18 0x0002
#define R01_SFORO_LSB20 0x0003
#define R01_SFORO_LSB24 0x0004
#define R01_SFORO_MSB 0x0005
#define R01_SFORO_MASK 0x0007
#define R01_SEL_SOURCE 0x0040
#define R01_SIM 0x0010
#define R02_PON_PLL 0x8000
#define R02_PON_HP 0x2000
#define R02_PON_DAC 0x0400
#define R02_PON_BIAS 0x0100
#define R02_EN_AVC 0x0080
#define R02_PON_AVC 0x0040
#define R02_PON_LNA 0x0010
#define R02_PON_PGAL 0x0008
#define R02_PON_ADCL 0x0004
#define R02_PON_PGAR 0x0002
#define R02_PON_ADCR 0x0001
#define R13_MTM 0x4000
#define R14_SILENCE 0x0080
#define R14_SDET_ON 0x0040
#define R21_MT_ADC 0x8000
#define R22_SEL_LNA 0x0008
#define R22_SEL_MIC 0x0004
#define R22_SKIP_DCFIL 0x0002
#define R23_AGC_EN 0x0001
struct uda1380_setup_data {
int i2c_bus;
unsigned short i2c_address;
int dac_clk;
void (* power)(int);
#define UDA1380_DAC_CLK_SYSCLK 0
#define UDA1380_DAC_CLK_WSPLL 1
};
#define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */
#define UDA1380_DAI_PLAYBACK 1 /* playback DAI */
#define UDA1380_DAI_CAPTURE 2 /* capture DAI */
extern struct snd_soc_dai uda1380_dai[3];
extern struct snd_soc_codec_device soc_codec_dev_uda1380;
#endif /* _UDA1380_H */
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